Based on
https://wiki.asterisk.org/wiki/display/AST/Module+Deprecation I think
you’re right when you say it shouldn’t be built by default
On Mon, 23 Aug 2021 at 15:15, Alexander Traud
wrote:
> While creating a minimal installation of the upcoming Asterisk 19 with
>
> ./configure
> make full
> su
Without any logs we can’t really help you…
Also a reminder that 13 is in security fix only mode and goes end of life
in October - it’s time to upgrade!
On Tue, 27 Apr 2021 at 06:13, Jaco Kroon wrote:
> Hi Rajesh,
>
> On 2021/04/27 05:46, Rajesh wrote:
>
> > We are using Asterisk 13.18.2. Asteri
Hi Josh,
Thanks for this! I have a question regarding this line in your email.
These changes allow the information regarding when a module was deprecated
and when it will be removed to be communicated to users in earlier branches.
Does this mean that app_foo being marked as deprecated in master
I was hoping you'd pipe in again Jared! Thank you!
On Fri, 2 Oct 2020, 20:20 Jared Smith, wrote:
> On Fri, Oct 2, 2020 at 11:50 AM Dan Jenkins wrote:
>
>> sorry, I thought I was agreeing with you :) we need to engage package
>> maintainers to potentially help ease the sh
el case, I don’t think anyone has
> missed it - it’s been far too long with two channels, which is confusing.
>
> There are propably a list of modules I would remove quickly if I was under
> Josh’s hat.
> /O
>
> On 2 Oct 2020, at 12:18, Dan Jenkins wrote:
>
> Ultimately
Ultimately whats stopping package maintainers from releasing
"asterisk-full" which still has all the deprecated modules enabled and
"asterisk" which follows the defaults? Nothing Other packages get
released in such a way so why not asterisk? I'm 100% not qualified to talk
about it because I
Completely agree with Sean on the "what's going to be deprecated" question
months before a cut. And I like the laid out plan that would be involved
for 2 year process of deprecation,default enabled no and then remove.
On Thu, 1 Oct 2020, 22:42 BJ Weschke, wrote:
> I don’t think anyone would have
rote:
> On Thu, Oct 1, 2020 at 3:56 PM Dan Jenkins wrote:
>
>> If there was an additional message attached to minor releases, does that
>> mean we can accelerate the steps?
>>
>> On the question of why I'm opposed to 4 years? 4 years is an eternity to
>> be
If there was an additional message attached to minor releases, does that
mean we can accelerate the steps?
On the question of why I'm opposed to 4 years? 4 years is an eternity to be
in limbo - we've already seen this with chan_sip - even though its
deprecated in 17, people still start using Aster
appen.
> On 10/1/20 12:25 PM, Joshua C. Colp wrote:
>
> On Thu, Oct 1, 2020 at 1:15 PM Dan Jenkins wrote:
>
>> Firstly, thank you Josh for trying to outline the process so that it's
>> something that can be followed and while I agree mostly with the steps...
>&
wouldnt be
enabled by default and ultimately thats what the changelog/upgrade.txt is
for isn't it?
4 years seems like a long time to get rid of something thats already been
decided isnt being looked after any more.
On Thu, Oct 1, 2020 at 5:27 PM Joshua C. Colp wrote:
> On Thu, Oct 1, 2
Firstly, thank you Josh for trying to outline the process so that it's
something that can be followed and while I agree mostly with the steps...
the fact that its going to take 4 years for a module to be moved from
"deprecated" to being removed just feels too long but understandable if
we're talkin
YES YES YES.
I never really got why it was done like it was...
I agree, 4-6 weeks is good. If someone's going to take the time to upgrade
to an 18 RC then they'll be looking to test it and give feedback etc etc so
you don't need a huge amount of time... just enough to actually action bug
fixes an
Hi George,
Funnily enough I saw this the other day and the client's Kamailio/RTPProxy
setup wasn't set up to handle the reinvite scenario afterwards. From what I
understand this is becoming more and more common. So in this case it was an
initial request to get the dialog going, once everything had
Thanks Josh! I'd seen the no dialplan needed but hadn't seen the move
functionality (woot!)
On Thu, 24 Oct 2019, 18:09 Joshua C. Colp, wrote:
> On Thu, Oct 24, 2019, at 1:59 PM, Dan Jenkins wrote:
> > Unfortunately I won't be at DevCon this year (I'll be on a plane
Unfortunately I won't be at DevCon this year (I'll be on a plane flying
from West to East US) - I will of course try to listen in etc but I'd like
to raise one new idea and another maintenance issue to be discussed at
DevCon.
Idea:
I find it tough to get people (even myself) to build actual produ
seems fair to me!
Looking forward to this George!
On Fri, Oct 18, 2019 at 1:57 PM George Joseph wrote:
> When we created the External Media addition to ARI we created an
> ExternalMedia object to be returned from the channels/externalMedia REST
> endpoint. This object contained the channel obj
Just caught up on this - makes sense to me.
DNS is important
transport - WSS to make it easy to just send binary down a WSS and have it
secure.
Dan
On Wed, Aug 7, 2019 at 4:41 PM Seán C. McCord wrote:
> Sounds like good reasoning to me.
>
> On Wed, Aug 7, 2019, 11:23 Joshua C. Colp wrote:
>
f
>>> coordination)
>>> - tear-down of listener
>>> - time-wise
>>> - after connection (what if nother ever connects?)
>>> - by command only
>>> - security
>>> - DoS vulnerability
>>>
>>> Technically, you
> since I multiplex the ARI amongst a wide range of processing nodes, but if
> you want to minimize additional routing and orchestration, you can't beat
> just simply using the same websocket that your control channel is on.
> That's about as coupled as it gets.
>
>
t;>
>>>
>>>
>>>
>>> *From:* asterisk-dev *On Behalf
>>> Of *Luca Pradovera
>>> *Sent:* Monday, July 22, 2019 3:12 AM
>>> *To:* Asterisk Developers Mailing List
>>> *Subject:* Re: [asterisk-dev] Audio to/from Asterisk
>>&
I
should be able to address an asterisk independently maybe this has just
proved that it needs to work both ways?
On Wed, Jul 24, 2019 at 4:19 PM George Joseph wrote:
>
>
>
> On Mon, Jul 22, 2019 at 2:01 AM Dan Jenkins wrote:
>
>> Also coming back to this with
;d talked about this mechanism at devcon?
Dan
On Sat, Jul 20, 2019 at 2:39 PM Dan Jenkins wrote:
> Just going to chime in and say I don't see a one way audio solution as
> enough and I'd worry that doing that would lead to "oh but only so many
> people need to chuck audi
Just going to chime in and say I don't see a one way audio solution as
enough and I'd worry that doing that would lead to "oh but only so many
people need to chuck audio in". This has been discussed at at least 3
devcons now.
On Thu, Jul 18, 2019 at 2:09 PM Seán C. McCord wrote:
> I certainly d
:thumbsup:
On Tue, Feb 5, 2019 at 11:03 AM Joshua C. Colp wrote:
> On Tue, Feb 5, 2019, at 5:27 AM, Dan Jenkins wrote:
> > so in this scenario a context will be created with "stasis-master"
> > and we can do all our magic if we want to; but its quite possible no
&
ot;, all calls will dial into that Stasis
> application. The context parameter shouldn't matter in this case, and with
> the addition of the new "move" REST API call, you can send your channel to
> any other active application. Overall, less work!
>
> Hopefully this answ
Ah so just to confirm - above in the thread theres a variable in the url
passed in called context so that you could have an app name of foo but a
context of bar and also an ari application without a context called alice
and therefore alice could still be addressable from dialplan. So do we want
the
your applications?
>>>
>>> On Wed, Dec 12, 2018 at 11:42 PM Anil Gupta
>>> wrote:
>>>
>>>> +1 to the context idea
>>>>
>>>> Something like *context = stasis:app_name* would be nice. I presume
>>>> this could be achieved by addi
Oh I do remember the context idea - which seemed like a good idea at the
time because of not having to change much internally
On Wed, Dec 12, 2018 at 7:07 PM Seán C. McCord wrote:
> Correction: when I said the "latter," I should have said the "third"
> option. The last option does not really w
On Wed, Dec 12, 2018 at 4:24 PM Ben Ford wrote:
> Hey all,
>
> We’re looking into AstriCon feedback and one of the things we want to
> touch on is ARI -- specifically, the ability to not have to create an
> extensions.conf in order to dial into Stasis. Before we start working on
> this, we’d lik
Thanks for the reminder Matt.
The minutes from devcon can be found here if anyone is interested -
https://wiki.asterisk.org/wiki/display/AST/astridevcon+2018+minutes
At devcon we talked about the growing need to be able to read audio from a
channel and be able to pump audio back into a channel to
Ha! Already informed them on Friday via other means. I'm told there is now
an IT ticket open
On Sun, 5 Aug 2018, 11:18 Alexander Traud, wrote:
> All asterisk.org (sub-) domains are secured by a SSL/TLS certificate from
> RapidSSL which chains up to the trust anchor "GeoTrust Global CA". That
> t
He means the rtpbleed issue I presume. And I would presume that its already
been patched. But I can't tell you for sure
https://rtpbleed.com/
On Tue, Sep 26, 2017 at 7:04 PM, Sean Bright wrote:
> On 9/26/2017 12:37 PM, sean darcy wrote:
>
>> Are there any RTP/RTCP leak issues with 15 RC1 ?
>>
>
On Thu, Sep 7, 2017 at 10:03 AM, Sylvain Boily
wrote:
>
>
> Le 2017-09-05 à 23:54, Matt Fredrickson a écrit :
>
>> On Wed, Aug 30, 2017 at 1:54 PM, Sylvain Boily
>> wrote:
>>
>>> Hello!
>>>
>>> We start a proof of concept based on thie review board
>>> (https://reviewboard.asterisk.org/r/4365/)
On Thu, Aug 17, 2017 at 11:11 PM, Matt Fredrickson
wrote:
> On Tue, Aug 15, 2017 at 1:15 PM, Dan Jenkins
> wrote:
> >
> >
> > On Wed, Aug 2, 2017 at 10:57 PM, Matt Fredrickson
> > wrote:
> >>
> >> It is with great pleasure I wish to inform you of
On Wed, Aug 2, 2017 at 10:57 PM, Matt Fredrickson
wrote:
> It is with great pleasure I wish to inform you of the first beta
> release of the new Asterisk 15 branch. It's a very exciting time to be
> a user of Asterisk! Asterisk 15 is arguably the biggest release of
> Asterisk that has happened in
On Tue, Aug 15, 2017 at 2:07 AM, George Joseph wrote:
>
>
> On Mon, Aug 14, 2017 at 1:04 PM, Dan Jenkins
> wrote:
>
>>
>>
>> On Tue, Aug 8, 2017 at 10:44 PM, George Joseph
>> wrote:
>>
>>>
>>>
>>> On Tue, Aug 8, 2017 at 1
On Tue, Aug 8, 2017 at 10:44 PM, George Joseph wrote:
>
>
> On Tue, Aug 8, 2017 at 1:15 PM, George Joseph wrote:
>
>> The option to use the bundled version of pjproject has been available
>> since January 2016 and is the only "supported" method of using pjproject
>> now. It's also the only effe
We've all been there and done that :D
On Wed, Aug 2, 2017 at 1:39 AM, George Joseph wrote:
> I messed up. The next time you pull one of the asterisk branches from git
> you may get an error like this...
>
> Switched to branch 'master'
> Your branch and 'origin/master' have diverged,
> and have
On Tue, Jun 6, 2017 at 6:28 PM, Mark Michelson
wrote:
> On 06/05/2017 03:17 PM, Matt Fredrickson wrote:
>
>> On Mon, Jun 5, 2017 at 2:31 PM, Joshua Colp wrote:
>>
>>> On Mon, Jun 5, 2017, at 04:21 PM, Mark Michelson wrote:
>>>
Hi folks,
For those of you following along at home, I
On Thu, Jan 19, 2017 at 9:09 AM, Dan Jenkins
wrote:
> Hi Sebastian,
>
> I haven't opened an issue in Asterisk yet - ran out of time yesterday -
> first thing on my list to do today.
>
> I have now tested the flag in jssip (jssip makes it extremely easy to test
> as it
>
> thanks
>
> best regards
>
>
> On Jan 18, 2017, 10:52 -0600, Dan Jenkins ,
> wrote:
>
> Hi All,
>
> I've been working with a company who utilise WebRTC using Asterisk behind
> Kamailio to connect browser users and their SIP infrastructure and just
> c
Hi All,
I've been working with a company who utilise WebRTC using Asterisk behind
Kamailio to connect browser users and their SIP infrastructure and just
came across an issue making/receiving calls in Chrome Canary and Chrome Dev.
Long story short; the issue is that rtcp-mux has now been set as r
On Tue, Dec 6, 2016 at 2:43 PM, George Joseph wrote:
> We just discovered that the PJSIPShowRegistrationsInbound command really
> only dumps all aors regardless of whether there's an inbound registration
> associated with it or not. Fixing this would mean a change to what this
> command returns
+1 to option 2
On Thu, Nov 17, 2016, 19:44 Bryant Zimmerman wrote:
>
>
> +1 to option 2.
>
> Thanks
>
> Bryant Zimmerman (ZK Tech Inc.)
> 616-855-1030 Ext. 2003
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.
On Mon, Nov 7, 2016 at 2:51 PM, Matthew Jordan wrote:
>
>
> On Mon, Nov 7, 2016 at 5:00 AM, Dan Jenkins
> wrote:
>
>>
>> On Fri, Nov 4, 2016 at 2:07 PM, Matt Fredrickson
>> wrote:
>>
>>> Hey All,
>>>
>>> I've been thinking
On Fri, Nov 4, 2016 at 2:07 PM, Matt Fredrickson wrote:
> Hey All,
>
> I've been thinking a lot about how working groups might work within
> the context of the Asterisk project. Here are a few guidelines that I
> have come up with governing working groups. Some of these guidelines
> come from t
On Wed, Oct 12, 2016 at 2:43 PM, Matt Fredrickson
wrote:
> I've been deliberately waiting to weigh in on this discussion -
> however, my thoughts are as follows:
>
> 1.) Marking a module as deprecated is *not* the same thing as moving
> it out of the code base. It simply means that there is no m
On Fri, Oct 7, 2016 at 3:10 AM, Rodrigo Ramírez Norambuena <
decipher...@gmail.com> wrote:
> On Thu, Oct 6, 2016 at 7:44 AM, Dan Jenkins
> wrote:
> >
> > Hi Rodrigo,
> >
>
> Hi Dan!,
>
>
> > [ .. ]
> >
>
> >
> > So if I under
On Wed, Oct 5, 2016 at 11:11 PM, Rodrigo Ramírez Norambuena <
decipher...@gmail.com> wrote:
> On Tue, 2016-10-04 at 17:09 -0500, Matt Fredrickson wrote:
> > Hey all,
>
>
> Hello!,
>
> > Welcome back to all of you who attended AstriDevCon. Thanks so much
> > for all of you that attended and gave s
On Thu, Oct 6, 2016 at 10:01 AM, marek cervenka wrote:
>
> Michael,
>>
>> What would be amazing is for you to tell us which features you are
>> missing (or were missing when you tried)
>>
>> If we start a working group around PJSIP migration then these points will
>> help drive that forward.
>>
>
On Wed, Oct 5, 2016 at 4:04 PM, Michael Ulitskiy
wrote:
> I am in the same situation. All my systems are business-critical and I'm
>
> yet to see a convincing argument to spend a lot of man power to migrate
> the systems.
>
> Yes, pjsip supposed to be more stable, but chan_sip in asterisk 11 (wit
On Tue, Oct 4, 2016 at 11:09 PM, Matt Fredrickson
wrote:
> Hey all,
>
> Welcome back to all of you who attended AstriDevCon. Thanks so much
> for all of you that attended and gave so much of your time to be able
> to contribute.
>
> One of the ideas proposed in AstriDevCon was to create the noti
On Wed, Oct 5, 2016 at 8:21 AM, Leandro Dardini wrote:
> Your analysis of the chan_sip/PJSIP is really great and I agree with you.
> Being a grey haired tech, I can check what drives similar changes in the
> latest 20 years. We moved from Netware networks to TCP/IP, we moved from
> Windows 3.11 t
On Sun, Aug 7, 2016 at 4:42 PM, Chad McElligott
wrote:
> Hi everyone,
>
> We recently merged a critical bug fix to node-ari-client master branch
> that only affected users of Node 6.3+ [1][2]. This fix was in the 'ws'
> library. By upgrading to the necessary version to address the bug, we
> consu
On Tue, Feb 9, 2016 at 4:26 PM, Matthew Jordan wrote:
>
>
> On Tue, Feb 9, 2016 at 3:56 AM, Dan Jenkins
> wrote:
>
>> Hi Everyone,
>>
>> I've been looking at how we can add proxy support (1) to the Node.js ARI
>> Client for the past couple of d
Hi Everyone,
I've been looking at how we can add proxy support (1) to the Node.js ARI
Client for the past couple of days and have hit a few issues which I'm sure
we'll be able to work out. But this has led me down the path of looking
into the current status of swagger.
Swagger recently donated it
On Tue, Apr 14, 2015 at 3:18 PM, Russell Bryant
wrote:
> On Tue, Apr 14, 2015 at 8:47 AM, Matthew Jordan
> wrote:
>
>> On Tue, Apr 14, 2015 at 2:15 AM, Olle E. Johansson
>> wrote:
>> > Can we possibly have different Subject: lines in e-mails for a commit
>> and for a review. Seems like
>> > eve
On Tue, Feb 3, 2015 at 7:40 AM, Vipul Rastogi
wrote:
> Today, I got Asterisk working with respoke channel code as well but on
> startup Asterisk connects (register=yes) with node.js server with HTTP GET
> which does not have following headers...
> Connection: Upgrade
> Upgrade: websocket
>
> hen
On Wed, Jan 21, 2015 at 2:36 AM, Vipul Rastogi wrote:
> Anybody tried asterisk connecting to Socket.io server as websocket client
> ? I am not getting websocket established.
>
> See below error, after successful 101 Switching Protocols
>
> res_http_websocket.c:576 __ast_websocket_read: WebSocket
On Wed, May 7, 2014 at 1:26 PM, Olle E. Johansson wrote:
>
> On 06 May 2014, at 19:13, Daniel McFarlane wrote:
>
> > On 05/06/2014 10:56 AM, Olle E. Johansson wrote:
> >> On 06 May 2014, at 16:11, Daniel McFarlane wrote:
> >>
> >>> Hi All,
> >>>
> >>> I've been working with Asterisk for 2.5 yea
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